similar to: False Zap answer problem (Again)

We have an asterisk box with the following configuration: - AMD Athlon XP 2400+ - 512 MB RAM - SUSE Linux 10.1 - a Digium card TDM400P with 3 FXO - another Digium card TDM400P with 4 FXS - asterisk 1.2.7.1 - zaptel 1.2.4 I already checked that those cards aren't sharing interrupts (by cat /proc/interrupts): 0: 14119786 XT-PIC timer 1: 10 XT-PIC i8042 2:

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

hello, I have thousands of DNS entries like BA15-1073 CNF:058373e2-f784-458f-8c7b-fbf76fa25694 in my Samba AD. I know that these records are conflicts between AD DC synchronisation, and that I can remove it safely. When I list the DNS with the command samba-tool dns query localhost DOMAIN @ ALL it appears that there is a \n between the host name (BA15-1073) and the rest of the name

Hi all, Has anyone made up custom ring tones for the Polycom SIP phones? We use different rings for different lines, but the ones it comes with are all very similar. In the interesting of sharing, here's one I made up for paging: <PAGE_BEEP se.pat.ringer.13.name="Page Beep" se.pat.ringer.13.inst.1.type="chord" se.pat.ringer.13.inst.1.value="12"

Working on a paging system for one of my sites and running into something I can't believe is this hard. In one of the zones, they want to have three different extensions ring over the pa system, using it as a loud ringer. Now the paging system does have a loud ringer built in and I can easily have it do a simultaneous ring, but all of the extensions will sound the same over the loud

I am using SugarCRM together with the asterisk plugin, which allows me to click a number, SugarCRM calls my extension then places the call when I pickup. I would like to have that extension auto-answer. I set it up as line 3 on my phone so normal calls do not get auto-answered. However, I have not been able to get this to work. Has anyone implented this? This is what I put in the config file

In ipmid.cfg I have: <G3INTERCOM se.rt.10.name="G3INTERCOM" se.rt.4.type="ring-answer" se.rt.4.timeout="1000" se.rt.10.ringer="7"/> In sip.cfg I have: <alertInfo voIpProt.SIP.alertInfo.1.value="G3INTERCOM" voIpProt.SIP.alertInfo.1.class="10"/> I set up a test extension: exten =>

Hi, I'm trying to set up my SNOM 200 with extensions, with different ringtones - but it doesn't seem to work. I've defined two extensions for it in Asterisk and in the SNOM 200 configuration. In the SNOM home>settings>SIP>Lines config page, I have set the ringer for the first extension to "ringer1", and "ringer6" for the second. In

Hi guys I'm using asterisk 1.4 and going on to Snom phones. I'm trying to add a sip header to make the Snom phone use a different ring tone on one particular incoming number. I have added the following to the dial plan of the incoming context +------+------------------+-------+----------+--------------+-------------------------+ | id | context | exten | priority | app

I am preparing to setup a system using Cisco 7940 and 7960's I have the 7.1 SIP firmware on them. One issue I have run into is how to silence the ringer if a call comes in and you don't want to take it. Many phones have a DND button. I know the 79XX has the DND in the menu but it is to cumbersome to go into the settings then phone preferences then the DND and select yes. Is there any other

I'm trying to change the ring tone on my 7960 from the dialplan. I've tried the example on the wiki but it doesn't seem to work. Something like: exten => 3010,1,SetVar(ALERT_INFO=<Bellcore-dr1>) ; selects Ringer exten => 3010,2,Dial(SIP/3010,15) I'm not sure what the Bellcore-dr1 ringer is supposed to be. I've tried replacing ALERT_INFO with another ring tone

Hi Andrew - > I have a need to be able to identify incoming calls based on some factor > (could be time of day, caller ID, dialed number, it doesn't matter.) -- > Assuming Asterisk can differentiate between the calls I want, how do I inform > the IP501? There are "only" three line appearances -- I can't simply just > ring a different appearance since there

1. How do you set the music on hold to work with asterisk. Right now when I place a call on hold the caller hears nothing. MOH works with all my other IP phones. 2. Ringer Volume. How do you set the ringer volume? So that it's set on reboot. Thanks

Hi, I have a customer with a definitely low-tech need: he has a noisy storeroom where he wants to hear the phones ringing so he can leave the storeroom and pick up the phone in his office. So all I need is a loud SIP ringer. Does this even exist? I know paging amplifiers exist, but that`s not what I need. Mike -------------- next part -------------- An HTML attachment was

I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to "terminate" POTS lines and use all SIP Phones? Due to circumstances, I end up with a 1u server that has no aux power connectors available. I "have to" use this server, so am considering abandoning

Hi list, Does anyone have any advice on the following: Incoming calls to our office come in on a SIP trunk. Since all our offices/desks are in close proximity, we would like just a single phone to ring when a call comes in instead of ringing every person's phone. Currently we've got this working by having all the phones in a callgroup/pickupgroup and incoming calls ring the

I noticed that the BudgeTone (I have the 102) with the latest firmware tries to download a file called cfg.txt (presumably the configuration) and a file called ring.bin (presumably a ringer) from the tftp server. The ring-in sound on the budgetone is the same as the ring-out sound and that is going to be confusing for users. I contacted GrandStream and was informed that both of these formats are

I am having an issue where outbound external calls. Calls made using an analog line (connected to an FXS) route correctly out the trunk (connected to an FXO). However, when I make a similar outbound call using a SIP phone the analog phone connected to the FXS rings. I was having this problem intermittently with a manual asterisk install -- a reboot would fix the problem. I am now giving

Folks, I have a TDM400 with an FXS module. I'm having some trouble ringing some phones attached to the device. I have a real el cheapo handset that cost me about 9UKP. It works fine, I plug it into the Wildcard and call the channel and it rings. I have tried plugging in my BT Diverse 3016 cordless phone but it will not ring, although I can call out from it. I have also tried my GPO 332L

Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable.